similar to: Attended Transfer with snom phones

Hi, I am using a number of snom190 phones, and an asterisk "gateway" server, and recently started experimenting with call transfers. The snom phones provide support for attended and un-attended call transfer, so I would rather use that than call-parking. I have found that un-attended transfer works fine, and that attended transfer works fine if the originating phone call is NON-SIP

With asterisk 1.2.17 and snom 320 version 6.2.23, if a user places a call on hold, then places another call and hangs up, the two calls are conferenced together. What is causing this and how to I stop it? Thanks!

Hello all, I am running into some problems with a snom 105 phone trying to do a attended transfer . Snom phones are connected to Asterisk. This does not work, it will only do a unattended transfer. I have downloaded the manual from snom and followed the instructions. Has anyone experienced the same problem ? any ideas how to solve the problem. thanks, Arne.

Hi everyone, Ive just installed a bunch of Snom 360s, and now having a NIGHTMARE of problems! Ive got a receponist phone with a extra sidecar on it. And when she gets 2+ calls and puts them on hold, when she goes to transfer them out the calls on hold get merged together. Somehow the calls on hold get merged and not to the extension needed!! Any help on this would be great guys, that would be

Hello, Before I dig into the SIP traces maybe someone has a clue: We have a few Snom phones, and the call-limit on the extension associated with them is set to 1 (so the other Snom phones can blink the relevant LED). When I call a SNOM phone (300 with the latest firmware) and the call is answered the phone is in "in use" state (verbose printing from Asterisk). When I hit the

Full_Name: Nicholas Boers Version: 1.8.1 OS: Windows XP Home Edition, Debian GNU/Linux 3.0 Submission from: (NULL) (24.68.84.10) mgp[2] specifies the axis labels' margin line relative to that of the axis line (mgp[3]). The `par' help does not document this behaviour. The following code demonstrates my observation: plot.new(); par(mgp = c(3, 2, 1)); box(); title(xlab =

Full_Name: Nicholas Boers Version: 1.8.1 OS: Windows XP Home Edition, Debian GNU/Linux 3.0 Submission from: (NULL) (24.68.84.10) mgp[2] specifies the axis labels' margin line relative to that of the axis line (mgp[3]). The `par' help does not document this behaviour. The following code demonstrates my observation: plot.new(); par(mgp = c(3, 2, 1)); box(); title(xlab =

I cannot get the transfer button to work on a Snom 190, I cannot get the # to work either. Any ideas? Regards Thorben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041207/5ac06747/attachment.htm

Hi all, I've programmed on a SNOM function key the destination "**77" which in my extension plan reprograms/toggles an asterisk DB variable which I use in the extension plan for some call routing decisions. I would like the SNOM extension light to permanently reflect the current toggle status of my application logic/asterisk DB variable. Is something like that possible to do in

Hello all! I have a few problems with Snom 320 phones: Problem A - Transfer out of Queues: We have a call center with some Snoms. We are using Queue and AgentCallbackLogin. As we run * 1.2.7.1 an agent could transfer a call out of the queue using the hold and transfer buttons on the Snom. This might have been the wrong way to do it all the time I found out later, but it worked. Now we upgraded

Hi, everyone: I'm in the processing of deciding what IP phones we should use with our Asterisk server, and I wanted to get feedback from the user community on the quality, reliability and ease of operation of Snom phones. What do you have to say about these phones? Are there other phones you'd suggest along with or instead of Snom? Thanks, -Stephen-

Hi, I'm using Grandstreams as the agents phone of a queue. Attended transfers in a normal situation (direct call to the extension) work fine, but when the agent has a queue call and tries to transfer it to another sip extension the called party is hung up. Transfer is to pick another line on the Grandstream, dial the other extension, wait for a pickup, and then press the transfer button +

Hi, I successfully installed asterisk 1.0 with Capi 0.35. In my pbx system I would like to use the atxfer function but is not included in the stable asterisk. Is there a way to include it in my version of asterisk: I did no used the last cvs because I can't compile the chan_capi .in it. :( Bye

I am having a hell of a time with transfers. First the Snom issues: The transfer button on the Snom 220 does not work. I have read about setting break key off in the advanced page of the web config but the Snom 220 has no such option. At the moment I am having to use the # transfer hack which makes this phone look really stupid to have buttons on it that cannot be used. Anyone know how to

On Thu, March 5, 2015 09:56, Ruben R?gels wrote: > > Hi again, > > I'm glad to hear that I provided a somehow useful answer. > > Unfortunatelly, I don't know these details. > If you wasn't lucky consulting the snom docs, maybe the snom support > can be helpful with information about the exact implementation > details. > > You also could use "sip

Hi, Has anyone here reprogrammed their Polycom features keys using sip/ipmid.cfg? If so I would be really grateful if someone could send me an example as I have tried various entries for hours now and don't seem to be getting anywhere. Any help appreciated. Kind regards Jamie Heckford Technical Consultant

Hi all, I have had quite a few customer complaints about attended transfer cutting off callers. The problem is when reception is busy she doesn't always wait for someone to answer the call, however hanging up a ringing transfer on attended also hangs up the caller. I have checked the scripts I don't *think* this is a dial plan error but if anyone has this working correctly on Asterisk

Hello, I am testing the Openstage phones from Siemens but I can not find a solution on how to update the caller-id after a successful attended transfer. Of course, I mean an attended transfer by using the phones functionality, not something defined in asterisks features.conf. Any idea on how to achieve this, or any technical document from Siemens on on how this is support to work would help.

On Thu, March 5, 2015 05:30, Ruben R?gels wrote: > > > Am 05.03.2015 um 01:09 schrieb James B. Byrne: >> I am trying to determine how the transfer button on the Snom-870 >> works >> with Asterisk. Is the ## special code employed as when it is >> entered >> through the handset or is the blind transfer through the phone >> function accomplished in a

Hi all, One of my users reports frequently hearing echo on her Snom 360 phone, even while talking to other Snom phones (via Asterisk) on the same LAN (i.e., all-digital low-latency connection). I can never reproduce it though, and swapping the phone didn't help. Has anyone else seen "mystery echo" on Snom phones? Any suggestions for debugging? On my own Snom 360, I sometimes